The present invention relates to an apparatus for speech synthesizing, in particular, relates to such an apparatus which provides excellent synthesized speech using a simple circuit and small capacity of memory.
ADPCM (Adaptive Differential Pulse Code Modulation) has been a known band compression technique for voice channel. The ADPCM system compresses the voice frequency signal by taking the difference between the actual signal level at time T.sub.2 and the predicted signal level calculated by the signal level at time T.sub.1. In the demodulation stage, the difference is sequentially accumulated to provide a voice signal in PCM form. The quantization step for the quantization of a voice signal depends upon an instantaneous voice signal level.
FIG. 1 shows a block diagram of a prior ADPCM reproducer, which reproduces an ADPCM signal to provide a voice signal. In FIG. 1, the reference numeral 1 is an input terminal, 2 is an adder, 3 is a multiplicator, 4 is an adder, 5 is a register, 6 is a table for converting ADPCM code L.sub.n to the quantization step size moving coefficient M.sub.n, 7 is a multiplicator, 8 is an amplitude limiter, 9 is a register, 10 is an output terminal of demodulated PCM signal and that output terminal 10 is coupled with a D-A converter (not shown) for converting digital voice signal to analog voice signal.
It is assumed that the ADPCM signal at the input terminal 1 is L.sub.n. The bias value 0.5 is added to the signal L.sub.n by the adder 2. The sum from the adder 2 is applied to the multiplicator 3 which provides the product of said sum and the output .DELTA..sub.n of the register 9. The output .DELTA..sub.n is called a quantization step size. The output q.sub.n of the multiplicator 3 is the differential reproduced value of the ADPCM code L.sub.n, and is expressed as follows. EQU q.sub.n =.DELTA..sub.n (L.sub.n +1/2) (1)
The differential reproduced value q.sub.n from the multiplicator 3 is added to the output x.sub.n of the register 5 in the adder 4, and the result x.sub.n+1 is stored in the register 5. Those results x.sub.n and x.sub.n+1 are reproduced PCM signals of a voice signal.
When ADPCM code has 3 bits, and reproduced PCM code has 12 bits, the compression ratio by using ADPCM code is 3/12=1/4.
Another system for the compression of a voice signal is the use of even symmetry of a voice signal. When the voice signal is even symmetrical with the samples 2N-1 as shown in FIG. 2, the following relations are satisfied, where f(T) is the amplitude of a voice signal.
f(T)=f((2N-1)T) PA1 f(2T)=f((2N-2)T) PA1 f((N-1)T)=f((N+1)T)
Therefore, it is possible to reproduce 2N-1 signals by using only N samples, and it should be noted that the information quantity of voice signal is even halved by using the even symmetrical nature of a voice signal. Accordingly, it has been desired to develop an ADPCM circuit to handle an even symmetrical voice signal.
Another system for the compression of a voice signal is the use of repetition of the similar waveforms. It has been known that the change of the transfer function of a vocal tract is very slow, and is constant during 30 msec, although it depends upon each person. In a vowel sound, the same waveforms are repeated in every pitch at least three times. Therefore, when the synthesized voice is repeated by three times, the voice is compressed to one-third.
However, a prior repetition compression system has the disadvantage that the level of the voice signal is constant during the repetition. FIG. 3 shows the explanation, in which FIG. 3(a) shows the waveform of the actual pronounciation, in which p shows the length of a pitch. FIG. 3(b) is the synthesized waveform by a prior art system, in which the pitches p.sub.1, p.sub.2 and p.sub.3 have the same level as one another, and the pitches p.sub.4, p.sub.5 and p.sub.6 have also the same level as one another. Accordingly, the change of the average power of the voice signal is stepwise as shown by the curve (b) of FIG. 3(c), although the average voice power of the actual speech changes smoothly as shown in the curve (a) of FIG. 3(c). The difference between the curves (a) and (b) of FIG. 3(c) causes the deterioration of the quality of the synthesized speech.
Accordingly, an improved speech synthesizer system which provides excellent quality speech with simple circuitry and small information quantity has been desired.